build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
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- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits.
- Add a native slin16 type, so that 16kHz codecs can translate without losing resolution.
(This doesn't affect anything immediately, until another codec has wb support.)
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r80661 | phsultan | 2007-08-24 13:42:46 +0200 (Fri, 24 Aug 2007) | 9 lines
Closes issue #10509
Googletalk calls are answered too early, which results in CDRs wrongly
stating that a call was ANSWERED when the calling party cancelled a
call before before being established.
We must not answer the call upon reception of a 'transport-accept' iq
packet, but this packet still needs to be acknowledged, otherwise the
remote peer would close the call (like in #8970).
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r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 lines
(closes issue #10437)
Reported by: haklin
Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak.
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r72125 | qwell | 2007-06-27 12:10:32 -0500 (Wed, 27 Jun 2007) | 4 lines
Don't modify a variable that we don't want modified. Make a copy of it instead.
Issue 10029, patch by phsultan with slight modifications by me (to remove needless casts).
Note: chan_jingle in trunk does not appear to have the same bug.
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r66157 | kpfleming | 2007-05-25 10:28:46 -0400 (Fri, 25 May 2007) | 3 lines
handle the GNUTLS library properly in the configure script and build system
don't build in OSP support unless we have found and are allowed to use SSL support
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r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line
This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered.
This also adds the mods from 1.4/r.61136;
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T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired.
Feels very much like the old Unix talk application.
This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.
A big thank you to everyone involved in this.
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r51328 | russell | 2007-01-19 13:08:25 -0600 (Fri, 19 Jan 2007) | 5 lines
Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are
actually the same thing. So, a digit would have been interpreted incorrectly
here. Since the channel driver will always have the begin and end callbacks
called for a digit, only support the button-down and button-up messages.
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
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Overall i think the previous change to ast_channel_alloc()
to close bug 7506 should have been done by defining
an ast_set_callerid_noevent() function that does the
setting without generating the event.
Lot less code duplication, and easier to handle.
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changes in both of the moving specs. Currently chan_gtalk is
compatible with the latest gtalk/libjingle version, and chan_jingle
needs a lot of work.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43185 f38db490-d61c-443f-a65b-d21fe96a405b
rizzo